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Q-the-STORM
24th October 2013, 00:34
Quoting madshi:FLAC 1.3.0 has just officially sanctioned what eac3to has been doing all along. So there's nothing for me to change.

thanks, good to know ^^

But if you just extract core audio, without actually encoding (e.g. EAC3 to AC3, DTS-HD to Core dts), I would suggest to just keep the delay.

why? eac3to just adds silent audioframes, it doesn't reencode the actual audiotrack, so it doesn't really matter if you're reencoding or not...

LigH
24th October 2013, 12:28
why? eac3to just adds silent audioframes, it doesn't reencode the actual audiotrack, so it doesn't really matter if you're reencoding or not...

If this feature is reliable, then I will revert my previous opinion.

People shall know, though, that audio frames usually have a different duration than video frame durations for all usual frame rates. Losslessly adding or removing whole audio frames at the beginning will probably not "fix" any arbitrary delay to the millisecond. Just to the next closest frame. Certainly below recognition threshold, though...

madshi
25th October 2013, 21:16
FWIW, I might get around to making a new build some time in the next couple of weeks. So if any bugs are known but not listed in the bug tracker yet, please someone add them to the tracker:

http://eac3to.bugs.madshi.net

Thanks. Just to make sure: The bug tracker is currently only open for bugs, not for feature requests.

sl1pkn07
25th October 2013, 21:31
madashi.

make work muxing mkv with eac3to in linux/wine is a bug (http://sl1pkn07.no-ip.com/paste/view/cec84682) or is a feature?

madshi
25th October 2013, 23:14
A feature.

therube
26th October 2013, 20:24
Hi. I've been using eac3to for a short bit of time now. Nice little program.
(I have not /yet/ read through this, eh, short, thread ;-).)

Most of what I've done so far are transcodes, with the final output to aac (m4a), low bitrates, 32-64 range, perhaps with an added delay (-###ms).

As far as I know Nero (neroAacEnc.exe v1.5.4) is the only AAC encoder you're supporting?
(I do not have Nero 7, or any "Nero" other then the command-line encoder.)

Reason I ask, my issue, is that I get "clipping", perhaps better described, when I jump through a clip, I'll typically hear a chirp (a pop) when I jump.
And I seem to recall a post, somewhere, or a wiki, that mentioned, something to that effect?

I do not get these artifacts if I use a higher bitrate (-quality=).

So, is it possible a wrong parameter, or conflicting parameter is being passed to Nero causing this?
Or can another aac encoder be used, fdkaac or whatever?

hubblec4
26th October 2013, 23:36
hi madshi

a short question.

Is it necessary that eac3to so deep/long scan a file(m2ts) or a BD-Structure when i want to demux only the chapters?

eac3to.exe "H:" 1) 1:chapter.txt

tebasuna51
27th October 2013, 09:52
...
Most of what I've done so far are transcodes, with the final output to aac (m4a), low bitrates, 32-64 range, perhaps with an added delay (-###ms)...

Maybe NeroAacEnc isn't the recommended encoder for these low bitrates. You can use other encoder than accept STDIN like input, for instance qaac, using the STDOUT of eac3to with this sintax:

eac3to input stdout.wav [eac3to-parameters] | qaac [qaac-parameters] -o output.m4a -

tebasuna51
29th October 2013, 09:20
10 posts about Clown BD moved to new thread Question about ClownBD v0.81 (http://forum.doom9.org/showthread.php?t=169646)

Richard1485
4th November 2013, 00:53
I didn't see this new option in 3.25 until now.

-downStereo

Does this mix the lfe channel or not?


a=wavsource()
flr = GetChannel(a, 1, 2)
fcc = GetChannel(a, 3, 3)
lrc = MixAudio(flr, fcc, 0.3694, 0.2612)
blr = GetChannel(a, 5, 6)
MixAudio(lrc, blr, 1.0, 0.3694)


If not and it's the same downmix as above, I would to thank madshi very much for adding it.

tebasuna51
4th November 2013, 02:19
@Jeff B
To mix lfe use also -mixlfe

See the downmix matrix here: http://forum.doom9.org/showthread.php?p=1600695#post1600695

With automatic normalization if mix go over 0 dB.

Richard1485
4th November 2013, 02:36
Thanks, tebasuna. Yes. I know how to mix lfe. I was just making sure that it wasn't included because I don't like it.

It seems that the stereo downmix is different from the custom one that you made for me for Behappy. I'll see how it sounds. :-)

EDIT: Automatic normalization is good.

LigH
4th November 2013, 09:13
Depends on the technique. Only 2-pass normalization can preserve the volume distribution over the whole play time. If it is done in one pass (like the "Automatic Gain Control" of "ancient" tape recorders), the loudness will change often.

tebasuna51
4th November 2013, 13:52
Of course is the 2-pass method.

phate89
7th November 2013, 03:52
Hi.. I'm using a lot lately eac3to to check recorded ts with transmission errors to detect where and how big are the errors. Even if the gap processing doesn't return a video and audio in sync (i actually don't care because i only need audio) i'm wondering how eac3to fill the gaps. It duplicates the frame near or simply insert a silence? And is it possible with some switch to avoid to create the gaps (that i don't need)?

Wile_E_Coyote
8th November 2013, 19:26
I've seen notes about adding -mixlfe which I usually do when I'm downmixing (-downDpl) but is there a way to keep the LFE channel when going from DTS to AC3 and just mixing the rears into the fronts?
The layout of the room I'm watching TV in isn't ideal (they rarely are) so I have to roll out the cables to connect the rear speakers of my 5.1 system and thought I'd see what just the front speakers plus my sub would sound like.
I invariably get the "clipping detected" message and second pass so wondered if this would help that too?

Richard1485
8th November 2013, 22:02
I've seen notes about adding -mixlfe which I usually do when I'm downmixing (-downDpl) but is there a way to keep the LFE channel when going from DTS to AC3 and just mixing the rears into the fronts?

You could run your 5.1 through Eac3to once to downmix.

Eac3to input.dts output.wavs -downDpl

Then run it through again to split it into individual wavs.

Eac3to input.dts output2.wavs

Be careful to give the output files from the second run a different name. After that, you could take the two wavs from the first run and the LFE from the second run and use WAV to AC3 encoder (http://www.videohelp.com/tools/EncWAVtoAC3) to make your 2.1. I always use the MUX Wizard to map my channels.

I don't see why this wouldn't work, but no doubt Tebasuna will correct me if I have overlooked something.

tebasuna51
9th November 2013, 03:26
@Wile_E_Coyote
I can't understand for what you want a downmix 5.1 -> 2.1, but you can test the Jeff B method.

When downmix the "clipping detected" message is normal (try add 5-6 channels in only 2) and means than automatic normalization works fine.

wiggaz
11th November 2013, 03:05
Hi everyone,
I have a problem with eac3to.
I'm trying to convert some dtshd files into AC3. I've done this a lot of times even between those files I'm talking about.
Here what I've done:
I've extracted those dtshd files and they are like this:
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Now, to convert them into AC3 I'm using this:
%_.ac3 -normalize
Well, after the 1st pass, eac3to get stuck at the .wav file and nothing happened then.
Here's the log:
eac3to v3.27
command line: "C:\Program Files (x86)\eac3to 3.27\eac3to.exe" "D:\audio.dtshd" "D:\audio.dtshd_.ac3"
-normalize -progressnumbers -log="C:\Program Files (x86)\eac3to 3.27\UsEac3To.log"
------------------------------------------------------------------------------
DTS Master Audio, 5.1 channels, 24 bits, 48kHz
(core: DTS, 5.1 channels, 24 bits, 1509kbps, 48kHz)
Decoding with ArcSoft DTS Decoder...
Writing WAV...
Creating file "D:\audio.dtshd_.ac3.pass1.wav"...
What can I do? What do you suggest?
I've already tried without the "-normalize" and nothing changed.
Thanks in advance.

Stereodude
11th November 2013, 03:45
Maybe NeroAacEnc isn't the recommended encoder for these low bitrates. You can use other encoder than accept STDIN like input, for instance qaac, using the STDOUT of eac3to with this sintax:

eac3to input stdout.wav [eac3to-parameters] | qaac [qaac-parameters] -o output.m4a -For whatever it's worth, I found piping from eac3to to qaac to be extremely slow. CPU usage of each process (qaac and eac3to) were in the single digits (~6-9%) on my Q6600, but eac3to writing to a wave file has no problem saturating a core (25%) with the same decoding settings. For me it's a lot faster to go to an intermediate wave file and feed that to qaac.

shinchiro
12th November 2013, 12:24
Does eac3to support stdin?

Richard1485
12th November 2013, 21:42
"C:\Program Files (x86)\eac3to 3.27\eac3to.exe" "D:\audio.dtshd" "D:\audio.dtshd_.ac3

I've found that Eac3to sometimes does not like multiple stops in output file-names. Try this.

eac3to "D:\audio.dtshd" "D:\audio.ac3"

tebasuna51
13th November 2013, 10:30
I'm trying to convert some dtshd files into AC3. I've done this a lot of times even between those files I'm talking about.
...
Well, after the 1st pass, eac3to get stuck at the .wav file and nothing happened then.

If work with some dtshd but not with other maybe the last one is corrupt. (I supose you have enough disk space for the wav file)

Does eac3to support stdin?

Nope.

manolito
23rd November 2013, 12:06
Sorry if this has been asked before - but this thread is really long...

Last time I looked eac3to could not handle AAC audio files, probably because I do not have Nero 7 installed. Since I have no intention to install Nero 7 I would like to ask if current versions of eac3to can handle AAC sources by using other decoders like ffdshow or faad.


Cheers
manolito

LigH
23rd November 2013, 12:14
From the help screen:

eac3to v3.27, freeware by madshi.net
...

Supported source formats:
...
(6) AAC audio

So with the help of the included libavcodec, probably yes.

the_weirdo
23rd November 2013, 12:22
Actually, eac3to still need "Nero Audio Decoder 2" (from Nero 7) to decode AAC. Currently, it can't use any other DirectShow decoders or libavcodec to do that.

Lenmaer
23rd November 2013, 13:08
I was wondering if it was possible in the future to have eac3to read avisynth scripts?
Tha would be great to make audio trims using eac3to.

therube
23rd November 2013, 14:21
eac3to could not handle AAC audio files, probably because I do not have Nero 7 installed
You do not need Nero 7.
All you need is Nero AAC Codec, Free.
http://www.nero.com/eng/company/about-nero/nero-aac-codec.php

Download (~2 MB), unzip & drop the .exe's into your eac3to directory.

manolito
23rd November 2013, 14:53
You do not need Nero 7.
All you need is Nero AAC Codec, Free.
http://www.nero.com/eng/company/about-nero/nero-aac-codec.php

Download (~2 MB), unzip & drop the .exe's into your eac3to directory.

Sorry, you are posting incorrect information...:mad:

Of course I have tried your recommendation before I posted my question here, and it sure DOES NOT WORK.
eac3to v3.27
command line: eac3to i:\test.mp4 i:\testac3
------------------------------------------------------------------------------
The format of the source file could not be detected. <ERROR>

So please, next time you post, please verify that your information is correct...:p


Cheers
manolito

tebasuna51
23rd November 2013, 17:32
eac3to never support mp4 container like input.
Also .m4a (only aac audio in mp4 container) is not supported.

.aac input (with ADTS headers for instance) is only supported with Nero 7 installed.

LigH
23rd November 2013, 19:02
:o Sorry for a wrong hope then.

therube
24th November 2013, 14:56
Something like this certainly does work, with eac3to & nero encoder:

eac3to.exe INPUT OUTPUT OPTIONS DELAY
eac3to.exe riot_INPUT.wav riot_OUTPUT.w.18-1350ms.speedup.m4a -quality=0.18 -speedup -1350ms



eac3to.exe SN.ac3 sn+2300.m4a +2300ms -quality=0.18



eac3to.exe ie.wav ie-speedup+2100.m4a -speedup -quality=0.75 +2100ms


eac3to does not take mp4 as a source.


Supported source formats:
(1) RAW, (L)PCM
(2) WAV (PCM, DTS and AC3), W64, RF64
(3) AC3, E-AC3
(4) DTS, DTS-ES, DTS-96/24, DTS-HD Hi-Res, DTS-HD Master Audio
(5) MP1, MP2, MP3 audio
(6) AAC audio
(7) MLP, TrueHD, TrueHD/AC3
(8) FLAC
(9) EVO/VOB/(M2)TS and MKV

Decoded audio data can be stored as / encoded to:
(1) RAW, (L)PCM
(2) WAV (PCM only), W64, RF64, AGM
(3) WAVs (multiple mono WAV files, PCM only)
(4) AC3
(5) DTS
(6) AAC
(7) FLAC

manolito
24th November 2013, 20:29
.aac input (with ADTS headers for instance) is only supported with Nero 7 installed.

@therube
Seems like you did not read my question... ;)

I asked if eac3to handles aac audio without Nero 7 installed, and tebasuna answered my question in a very clear way. So the eac3to feature list is a little bit misleading.


Cheers
manolito

LigH
25th November 2013, 11:19
"Misleading", indeed. If it is written in such a general form like

(6) AAC audio

the "average user" would probably assume that it is generally supported without specific prerequisites. But after reading the longer explanation by tebasuna51, that is obviously not true. Just not every user will have read this post (http://forum.doom9.org/showthread.php?p=1654965#post1654965).

Therefore I would recommend to include such unexpected prerequisites in the help text:

(6) AAC audio (ADTS only, requires Nero 7)

At least as long as eac3to does not use LAV or L-SMASH "completely" as alternative splitter/decoder.

shh
26th November 2013, 11:50
Hi!
How do I extract the core AC3 stream of a combined E-AC3/AC3-stream?
My file looks exactly like a combined DTS-MA/DTS-core stream, but eac3to always want's to convert the file to AC3:
C:\eac3to\eac3to.exe german.ac3 out.ac3 -core
E-AC3, 7.1 channels, 1:44:16, 896kbps, 48kHz
(core: AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz)
Extracting E-AC3 core...
Decoding with libav/ffmpeg...
Remapping channels...
Encoding AC3 <640kbps> with libAften...
Creating file "coreac3.ac3"...
Clipping detected, a 2nd pass will be necessary.
-------^C

Snowknight26
26th November 2013, 15:54
http://forum.doom9.org/showthread.php?t=169776

shh
26th November 2013, 19:50
http://forum.doom9.org/showthread.php?t=169776
Oops, thank you very much! :eek:

I got it to work:
You have to create an output.ec3, though the result is an AC3 actually. So you also have to rename your .ec3 to .ac3 afterwords.
Hmm. This should be considered as a bug: I'm creating a real AC3, no E-AC3. Why should I name my output ec3? And, additionally, where's the doc to this secret .ec3-switch? I never read about an ec3 file before...
For .dtsma -> .dts -core the engine works correctly.

Perhaps I should've used full paths in my first post :rolleyes:
(no audio issue for me)
D:\monster>C:\eac3to\eac3to.exe german.ec3 coreac3.ec3 -core
E-AC3, 7.1 channels, 1:44:16, 896kbps, 48kHz
(core: AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz)
Extracting E-AC3 core...
Creating file "coreac3.ec3"...
eac3to processing took 5 seconds.
Done.

D:\monster>C:\eac3to\eac3to.exe coreac3.ec3
AC3, 5.1 channels, 1:44:16, 512kbps, 48kHz

Snowknight26
26th November 2013, 20:42
I wouldn't call it a bug per se but more of an unintended consequence of the relationship between E-AC3 and AC3, the fact that eac3to can natively encode AC3 (which it tries to do by default), and that eac3to only recently added support for extracting the E-AC3 core (from Blu-ray tracks only).

You don't have to name it .ec3.. you could choose .eac3 because, again, with .ac3 eac3to will try to transcode it by default, regardless of the -core switch. DTS, however, cannot be encoded by eac3to without additional software, hence the reason the -core switch works as you want it to.

Try the following and you'll see what I mean:

eac3to file.dtshd core.dtshd -core

As expected, the output is still the 'core' DTS track but with the .dtshd extension.

LigH
27th November 2013, 08:11
The fact that the work mode of eac3to is determined mainly by the filename extension of the output, appears to be a weakness in this case.

Snowknight26
27th November 2013, 16:31
The fact that the work mode of eac3to is determined mainly by the filename extension of the output, appears to be a weakness in this case.

True. Maybe when (or should I say 'if') eac3to starts using ffmpeg for processing/conversion will it be a good idea to switch to its command line arguments style of specifying formats (-acodec, -vcodec, -f, etc.).

Q-the-STORM
1st December 2013, 20:08
is there a way to force eac3to to encode AC3 with libav instead of aften? Justin Ruggles himself said that libav is better than aften and aften shouldn't be used anymore... he should know, since he developed aften and is now involved in libav... is there a way to force eac3to to use it? is there any reason not to use libav by default?

also on a similar matter, any way to get eac3to to use the DTS-HD Master Audio Suite (if you own it) for encoding DTS? would definitely be an improvement to surcode...

filler56789
1st December 2013, 23:47
is there a way to force eac3to to encode AC3 with libav instead of aften?

I think you can tell eac3to to write to stdout and pipe into ffmpeg or avconv...

also on a similar matter, any way to get eac3to to use the DTS-HD Master Audio Suite (if you own it) for encoding DTS? would definitely be an improvement to surcode...

If you donate a considerable amount of $$$$$ to madshi, he may think about that feature request ^_^ Seriously, the Master Audio Suite is a Java application, and unless I'm very misinformed, madshi is not a Java enthusiast :)

maxmercy
5th December 2013, 02:13
I have a 2 channel track that when played through an AVR will upmix to LCR and a single surround channel. Can eac3to 'upmix' this to individual L,C,R,S wavs?

JSS

DoctorM
5th December 2013, 08:05
maxmercy - I've previously requested that they take a look at a DPLII upmix option. There didn't seem to be much interest.

tebasuna51
5th December 2013, 10:15
Can eac3to 'upmix' this to individual L,C,R,S wavs?

Nope, eac3to can't do this upmix.

You can try Foobar2000 and the plugin foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235)

LigH
5th December 2013, 10:51
I am not certain, but I believe to remember that Selur was interested in such matrices too, it may be avaliable in his "Hybrid" converter.

Q-the-STORM
6th December 2013, 01:26
I think you can tell eac3to to write to stdout and pipe into ffmpeg or avconv...


thanks, I'll try that...

other question still remains, why is libav not used by default in eac3to?

maxmercy
6th December 2013, 12:59
Nope, eac3to can't do this upmix.

You can try Foobar2000 and the plugin foo_dsp_fsurround (http://www.hydrogenaudio.org/forums/index.php?showtopic=52235)

Thanks for the tip. Is there a way that foobar can write a .w64 or WAVEX after upmixing the stereo/prologic source?

JSS

torturesauce
9th December 2013, 12:40
The "eac3to and more" GUI does not work well at my Windows 8.1. It looks blown out of proportion and many menus/buttons are hidden. I tried disabling DPI scaling and it still doesn't look right. Is there any way to fix it? Or are there any other GUIs for eac3to that have the same amount of options?

the_weirdo
9th December 2013, 14:30
The "eac3to and more" GUI does not work well at my Windows 8.1. It looks blown out of proportion and many menus/buttons are hidden. I tried disabling DPI scaling and it still doesn't look right. Is there any way to fix it? Or are there any other GUIs for eac3to that have the same amount of options?

I don't actually use it but I think you can give this GUI (http://forum.doom9.org/showthread.php?t=145574) a try.